• Proceedings of the Korea Multimedia Society Conference
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• 2003.11b
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• pp.503-507
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• 2003

사용자의 고화질 요구와 사업자의 차별화된 서비스 제공 노력의 결과로 기존의 MPEG-4 기반이 아닌 고화질 전용의 MPEG-2 화질을 사용하는 VOD 서비스가 새로운 대안으로 제시되고 있다. MPEG-2 비디오는 높은 네트워크 대역폭을 요구하는 단점이 있는 반면, 사용자에게 양질의 화질을 제공할 수 있으며 표준의 사용으로 컨텐츠 유지. 보수에 유리하다. 본 논문에서는 상용 스트리밍 서버인 nCUBE 서버와 연동하여 MPEG-2 TS 데이터를 스트리밍 하는 VOD 시스템에 대하여 설명한다. VOD 제어 프로토콜로 RTSP(Real Time Streaming Protocol)를 사용하였으며, 스트림 전송 프로토콜로 UDP/IP 방식을 사용하였다. 지원하는 VCR 기능으로는 FF, RW, STOP. Pause가 있다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2022.05a
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• pp.524-526
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• 2022

Considering QoS only considering real-time multimedia service, it is possible to adjust the number of terminals and ensure them appropriately, but this study considers complex services considering real-time multimedia service and general data service. Since the amount of physical network resources is limited, the guarantee of the desired QoS can not be achieved unless the appropriate CAC is done. However, given the traffic profile and QoS spec of the entire network resource and the current service being provided, and the traffic profile and QoS spec of the newly requested service, it is quite difficult to determine exactly whether the new service request is acceptable from this. To do this, it is necessary to study in various directions from mathematical analysis to various simulations and statistical research based on data obtained from actual network operation.

• Journal of KIISE:Information Networking
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• v.37 no.6
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• pp.483-488
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• 2010

Due to the prevalence of various mobile devices and wireless broadband networks, there has been a significant increase in interest and demand for multimedia streaming services. Moreover, the user can service the participatory video broadcasting service in the mobile device and it can be used to deliver the real-time news and more variety information in the user side. Live multimedia service of user participation should consider not only the video quality but also the delay bounds and continuity of video playback for improving the user perceived QoS (Quality of Service) of streaming service. In this paper, we propose an adaptive rate control scheme, called DeBuG (Delay Bounds Guaranteed), to guarantee the delay bounds and continuity of video playback for the real-time streaming in mobile devices. In order to provide those, the proposed scheme has a quality adaptation function based on the transmission buffer status and network status awareness. It also has a selective frame dropper, which is based on the media priority, before the transmission video frames. The simulation results demonstrate the effectiveness of our proposed scheme.

• Journal of Digital Convergence
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• v.19 no.4
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• pp.251-257
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• 2021

With the development of new media, great changes are taking place in the way people get information. The change is the use of video content that can deliver content in a more three-dimensional way than words or photos. After 2016, the number of live video streaming content providers and users has increased. In this paper the write takes the 1 personal live video streaming content as the research object. And the write takes live video streaming content on YouTube live or Douyu TV as a research example. In this paper, the writer analyzes the digital information content in the live video streaming case. And the writer expounds the necessity of these visual information and the characteristics of real-time live video streaming content. Especially since 2020, because of the influence of the COVID-19, the live video streaming industry has begun to combine with the traditional industry. It is expected that the integration of digital cutting-edge technology and live video streaming will not only provide diversity in the content, but also create more social value for the video content consumption culture. Therefore, The writer thinks it is necessary to conduct in-depth research on the social responsibility of real-time live content in the future.

• 한국정보통신설비학회:학술대회논문집
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• 2008.08a
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• pp.584-587
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• 2008

In this paper, we propose a real time video codec adapter for enabling video communications with terminals having a codec which is different from each other. When multimedia services are playing with an office service phone such as a video phone or software phone which has video capability, each terminal is not being considered to have optimized video or voice codec. So when a video phone with only one type of video codec is used in the video streaming service which requires another type of codec, the streaming service is not successful without codec transformation. The real time codec adapter in this paper provides a real time code transformation which enables communication services such as video conferencing between terminals which have different codec.

• Proceedings of the Korea Multimedia Society Conference
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• 2002.11b
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• pp.767-770
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• 2002

본 논문에서는 RTSP 프로토콜을 제어하기 위한 서버 시스템을 IOCP 기반의 Multi-Thread 기법을 이용하여 구현하는 방법을 소개한다. 다수의 클라이언트에 대한 응답을 Thread로 구성하는 부분에서 Multi-Threading을 이용함으로써 수행 속도를 높이고 Winsock2에서 제공하는 IOCP(T/O Completion Port)를 이용하여 견고하고 확장이 용이한 RTSP(Real Time Streaming Protocol) 스트리밍 서버를 개발하였다.

• Proceedings of the CALSEC Conference
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• 2001.08a
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• pp.631-635
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• 2001

The Synchronized Multimedia Integration Language (SMIL) is a declarative markup language based on the eXtensible Markup Language (XML) to define a set of markup tags for synchronizing the timing and positioning relationships between multimedia objects. SMIL makes authoring of TV-like multimedia presentations on the Web easier for applications such as electronic commerce. We present the design and implementation of a JAVA-based SMIL player, which processes different types of media objects using multiple threads. Moreover, its cache engine detects the media type and allocates the proper cache memory for the corresponding media object.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.4B
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• pp.227-233
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• 2007

According to the recent development of network technology, the demands of users are diversified and the needs of multimedia traffic are increasing. In general, UDP(User Datagram Protocol) traffic is used to transport multimedia data, which satisfied the real-time and isochronous characteristics. UDP traffic competes with TCP traffic and incur the network congestion. However, TCP traffic performs network congestion control but does not consider the receiver's status. Thus, it is not appropriate in case of streaming services. In this paper, we solve a fairness problems and proposed a network algorithm based on RTP/RTCP(Real-time Transport Protocol/Realtime Transport Control Protocol) in view of receiver status. The POBA(Priority Ordered Buffer Algorithm), which applies priorities in the receiver's buffer and networks, shows that it provides the appropriate environment for streaming services in view of packet loss ratio and buffer utilization of receiver's buffer compared with the previous method.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.15 no.8
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• pp.3025-3047
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• 2021

This paper presents the latest Ethernet standardization of in-vehicle network and the future trends of automotive Ethernet technology. The proposed system provides design and optimization algorithms for automotive networking technology related to AVB (Audio Video Bridge) technology. We present a design of in-vehicle network system as well as the optimization of AVB for automotive. A proposal of Reduced Latency of Machine to Machine (RLMM) plays an outstanding role in reducing the latency among devices. RLMM's approach to real-world experimental cases indicates a reduction in latency of around 41.2%. The setup optimized for the automotive network environment is expected to significantly reduce the time in the development and design process. The results obtained in the study of image transmission latency are trustworthy because average values were collected over a long period of time. It is necessary to analyze a latency between multimedia devices within limited time which will be of considerable benefit to the industry. Furthermore, the proposed reliable camera and video streaming through optimized AVB device settings would provide a high level of support in the real-time comprehension and analysis of images with AI (Artificial Intelligence) algorithms in autonomous driving.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.25 no.4
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• pp.581-587
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• 2021

Real-time video streaming scheme for multimedia content delivery and remote conference services is one of technologies that are significantly sensitive to data transmission delay. Recently, because of COVID-19, real-time video streaming contents for the services are significantly increased such as personal broadcasting and remote school class. In order to support the services, there is a growing emphasis on low transmission delay and secure content delivery, respectively. Therefore, our research proposed a packet aggregation algorithm to reduce the transmission delay of selectively encrypted video transmission for real-time video streaming services. Through the application of the proposed algorithm, the selectively encrypted video framework can control the amount of MPEG-2 TS packets for low latency transmission with a consideration of packet priorities. Evaluation results on testbed show that the application of the proposed algorithm to the video framework can reduce approximately 11% of the transmission delay for high and low resolution video.